Convert Planar Audio to Packed Audio in FFmpeg

This guide provides a straightforward tutorial on how to convert planar audio formats to packed audio formats using FFmpeg. You will learn the difference between these two audio layouts, the specific command-line options required for conversion, and how to verify the output format of your audio files.

Understanding Planar vs. Packed Audio

In digital audio, channel layout determines how multi-channel data (like stereo) is stored: * Packed (Interleaved) Formats: Channel samples are stored consecutively. For a stereo file, the data is structured as L-R-L-R-L-R. In FFmpeg, these formats do not have a suffix (e.g., s16, flt, s32). * Planar Formats: Each channel’s data is stored in its own separate plane or block. The data is structured as L-L-L... followed by R-R-R.... In FFmpeg, these formats always end with a p suffix (e.g., s16p, fltp, s32p).

Method 1: Using the -sample_fmt Option

The simplest way to convert planar audio to packed audio is by using the -sample_fmt (or -pix_fmt equivalent for audio) flag. This tells the audio encoder to output a specific sample format.

For example, to convert an audio file with planar float samples (fltp) to packed signed 16-bit PCM (s16), use the following command:

ffmpeg -i input.wav -c:a pcm_s16le -sample_fmt s16 output.wav

In this command: * -c:a pcm_s16le selects the 16-bit PCM encoder. * -sample_fmt s16 forces FFmpeg to use the packed 16-bit sample format instead of the planar s16p.

Method 2: Using the aformat Filter

If you are performing complex filtering or if the encoder requires a specific input format, you can use the aformat filter to explicitly define the sample format before the audio reaches the encoder.

ffmpeg -i input.mp4 -af "aformat=sample_fmts=s16" -c:a pcm_s16le output.wav

The -af "aformat=sample_fmts=s16" filter forces the audio stream to be converted to the packed s16 format during the filtering stage.

Important Codec Considerations

Not all audio encoders support packed formats. For example, the default AAC encoder in FFmpeg (aac) natively operates on planar float (fltp) samples. If you force a packed format like flt on the AAC encoder, FFmpeg will automatically convert it back to fltp to encode the file.

To keep your audio in a packed format, you must output to a codec that supports packed data, such as PCM (pcm_s16le for s16, pcm_f32le for flt) or MP3.

How to Verify the Audio Format

To check whether your output file is using a packed or planar format, use ffprobe:

ffprobe -v error -show_entries stream=sample_fmt -of default=noprint_wrappers=1 input.wav

If the output is s16 or flt, your audio is successfully packed. If the output is s16p or fltp, it is still planar.